Moving/renaming webrtc/common.h.

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 172 matching lines @@
     return channels_[channel]->max_encoding_bandwidth;
   }
   int GetSendCNPayloadType(int channel, bool wideband) {
     return (wideband) ?
         channels_[channel]->cn16_type :
         channels_[channel]->cn8_type;
   }
   void set_fail_create_channel(bool fail_create_channel) {
     fail_create_channel_ = fail_create_channel;
   }
-  int AddChannel(const webrtc::Config& config) {
+  int AddChannel(const webrtc::AudioCodingModule::Config& acm_config) {
     if (fail_create_channel_) {
       return -1;
     }
     Channel* ch = new Channel();
     auto db = webrtc::acm2::RentACodec::Database();
     ch->recv_codecs.assign(db.begin(), db.end());
-    if (config.Get<webrtc::NetEqCapacityConfig>().enabled) {
-      ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity;
-    }
-    ch->neteq_fast_accelerate =
-        config.Get<webrtc::NetEqFastAccelerate>().enabled;
+    ch->neteq_capacity = acm_config.neteq_config.max_packets_in_buffer;
+    ch->neteq_fast_accelerate = acm_config.neteq_config.enable_fast_accelerate;
     channels_[++last_channel_] = ch;
     return last_channel_;
   }
 
   int GetNumSetSendCodecs() const { return num_set_send_codecs_; }
 
   int GetAssociateSendChannel(int channel) {
     return channels_[channel]->associate_send_channel;
   }
 
@@ skipping 15 matching lines @@
     inited_ = false;
     return 0;
   }
   webrtc::AudioProcessing* audio_processing() override {
     return &audio_processing_;
   }
   webrtc::AudioDeviceModule* audio_device_module() override {
     return nullptr;
   }
   WEBRTC_FUNC(CreateChannel, ()) {
-    webrtc::Config empty_config;
-    return AddChannel(empty_config);
+    return AddChannel(webrtc::AudioCodingModule::Config());
   }
-  WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) {
-    return AddChannel(config);
+  WEBRTC_FUNC(CreateChannel,
+              (const webrtc::AudioCodingModule::Config& acm_config)) {
+    return AddChannel(acm_config);
   }
   WEBRTC_FUNC(DeleteChannel, (int channel)) {
     WEBRTC_CHECK_CHANNEL(channel);
     for (const auto& ch : channels_) {
       if (ch.second->associate_send_channel == channel) {
         ch.second->associate_send_channel = -1;
       }
     }
     delete channels_[channel];
     channels_.erase(channel);
@@ skipping 321 matching lines @@
   webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone;
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_